Move responsibility for RTP header extensions on video receive.

webrtc/video/video_send_stream_tests.cc

 /*
  *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 #include <algorithm>  // max
@@ skipping 1433 matching lines @@
       call_ = sender_call;
     }
 
     void ModifyVideoConfigs(
         VideoSendStream::Config* send_config,
         std::vector<VideoReceiveStream::Config>* receive_configs,
         VideoEncoderConfig* encoder_config) override {
       send_config->rtp.extensions.clear();
       send_config->rtp.extensions.push_back(RtpExtension(
           RtpExtension::kTransportSequenceNumberUri, kExtensionId));
+#if 0
       (*receive_configs)[0].rtp.extensions = send_config->rtp.extensions;
+#endif
       (*receive_configs)[0].rtp.transport_cc = true;
     }
 
     void ModifyAudioConfigs(
         AudioSendStream::Config* send_config,
         std::vector<AudioReceiveStream::Config>* receive_configs) override {
       send_config->rtp.extensions.clear();
       send_config->rtp.extensions.push_back(RtpExtension(
           RtpExtension::kTransportSequenceNumberUri, kExtensionId));
       (*receive_configs)[0].rtp.extensions.clear();
@@ skipping 1865 matching lines @@
     rtc::CriticalSection crit_;
     uint32_t max_bitrate_bps_ GUARDED_BY(&crit_);
     bool first_packet_sent_ GUARDED_BY(&crit_);
     rtc::Event bitrate_changed_event_;
   } test;
 
   RunBaseTest(&test);
 }
 
 }  // namespace webrtc